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CME SIP- Disable SIP ALG OUT TO

speakip
Beginner
Beginner

Running CME 12.0 on (CISCO2901) -----> SIP Trunk---PSTN
CISCO2901 is also serving as GW for the network. Outbound calls sending random port to PSTN. Can someone show me how to Disable SIP ALG in this scenario?
Used these with no luck:

int g0/0
ip nat outside
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060
ip nat inside source list 1 interface GigabitEthernet0/1 overload
!
access-list 1 permit 10.10.8.0 0.0.0.255

++output of SIP calls
2022-09-07 07:59:00 -0400 : 91.x.3.43:59959 -> 104.219.163.73:5060
REGISTER sip:px3.nexvortex.com:5060 SIP/2.0
CSeq: 309 REGISTER
Contact: <sip:00@91.x.3.43:5060>

1 Accepted Solution

Accepted Solutions

Why didn't you say already at the beginning, that it's all about audio-issues?!
If you have audio problems, it has nothing to with SIP ALG or from which ports the SIP messages are sent from ... back to basics I would suggest!

Have you taken a packet capture on the router-interface towards provider and checked if you received / transmit packets?
And no, you cannot pin down all RTP traffic to just one port for every call
In voice-service-voip you can set the RTP port range (which the router is using) with the command rtp-port range <start-port> <end-port>

But you have to check, where the packets are dropped.

View solution in original post

3 Replies 3

b.winter
Advocate
Advocate

This is completely normal behavior, that the source port is a random one.
If you use SIP over TCP, then the port of the established TCP session is used. And in my opinion, SIP ALG has nothing to do with it.

If CUBE should always use the same TCP port, then use the command "conn-reuse" for SIP over TCP or "connection-reuse via-port" for SIP over UDP.

speakip
Beginner
Beginner

It didn't work, still sending random port to the SIP provider causing one way audio issue.... Any thoughts?

Why didn't you say already at the beginning, that it's all about audio-issues?!
If you have audio problems, it has nothing to with SIP ALG or from which ports the SIP messages are sent from ... back to basics I would suggest!

Have you taken a packet capture on the router-interface towards provider and checked if you received / transmit packets?
And no, you cannot pin down all RTP traffic to just one port for every call
In voice-service-voip you can set the RTP port range (which the router is using) with the command rtp-port range <start-port> <end-port>

But you have to check, where the packets are dropped.

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